Move src/voice to src/audio for better naming
Many of these files are used by Audio and Voice messages. Fixes https://github.com/vector-im/element-web/issues/18131
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21 changed files with 14 additions and 14 deletions
349
src/audio/VoiceRecording.ts
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349
src/audio/VoiceRecording.ts
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/*
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Copyright 2021 The Matrix.org Foundation C.I.C.
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Licensed under the Apache License, Version 2.0 (the "License");
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you may not use this file except in compliance with the License.
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You may obtain a copy of the License at
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http://www.apache.org/licenses/LICENSE-2.0
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Unless required by applicable law or agreed to in writing, software
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distributed under the License is distributed on an "AS IS" BASIS,
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WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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See the License for the specific language governing permissions and
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limitations under the License.
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*/
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import * as Recorder from 'opus-recorder';
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import encoderPath from 'opus-recorder/dist/encoderWorker.min.js';
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import { MatrixClient } from "matrix-js-sdk/src/client";
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import MediaDeviceHandler from "../MediaDeviceHandler";
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import { SimpleObservable } from "matrix-widget-api";
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import EventEmitter from "events";
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import { IDestroyable } from "../utils/IDestroyable";
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import { Singleflight } from "../utils/Singleflight";
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import { PayloadEvent, WORKLET_NAME } from "./consts";
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import { UPDATE_EVENT } from "../stores/AsyncStore";
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import { Playback } from "./Playback";
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import { createAudioContext } from "./compat";
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import { IEncryptedFile } from "matrix-js-sdk/src/@types/event";
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import { uploadFile } from "../ContentMessages";
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import { FixedRollingArray } from "../utils/FixedRollingArray";
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import { clamp } from "../utils/numbers";
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const CHANNELS = 1; // stereo isn't important
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export const SAMPLE_RATE = 48000; // 48khz is what WebRTC uses. 12khz is where we lose quality.
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const BITRATE = 24000; // 24kbps is pretty high quality for our use case in opus.
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const TARGET_MAX_LENGTH = 120; // 2 minutes in seconds. Somewhat arbitrary, though longer == larger files.
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const TARGET_WARN_TIME_LEFT = 10; // 10 seconds, also somewhat arbitrary.
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export const RECORDING_PLAYBACK_SAMPLES = 44;
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export interface IRecordingUpdate {
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waveform: number[]; // floating points between 0 (low) and 1 (high).
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timeSeconds: number; // float
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}
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export enum RecordingState {
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Started = "started",
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EndingSoon = "ending_soon", // emits an object with a single numerical value: secondsLeft
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Ended = "ended",
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Uploading = "uploading",
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Uploaded = "uploaded",
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}
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export interface IUpload {
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mxc?: string; // for unencrypted uploads
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encrypted?: IEncryptedFile;
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}
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export class VoiceRecording extends EventEmitter implements IDestroyable {
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private recorder: Recorder;
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private recorderContext: AudioContext;
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private recorderSource: MediaStreamAudioSourceNode;
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private recorderStream: MediaStream;
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private recorderWorklet: AudioWorkletNode;
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private recorderProcessor: ScriptProcessorNode;
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private buffer = new Uint8Array(0); // use this.audioBuffer to access
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private lastUpload: IUpload;
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private recording = false;
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private observable: SimpleObservable<IRecordingUpdate>;
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private amplitudes: number[] = []; // at each second mark, generated
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private playback: Playback;
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private liveWaveform = new FixedRollingArray(RECORDING_PLAYBACK_SAMPLES, 0);
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public constructor(private client: MatrixClient) {
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super();
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}
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public get contentType(): string {
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return "audio/ogg";
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}
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public get contentLength(): number {
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return this.buffer.length;
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}
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public get durationSeconds(): number {
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if (!this.recorder) throw new Error("Duration not available without a recording");
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return this.recorderContext.currentTime;
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}
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public get isRecording(): boolean {
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return this.recording;
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}
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public emit(event: string, ...args: any[]): boolean {
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super.emit(event, ...args);
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super.emit(UPDATE_EVENT, event, ...args);
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return true; // we don't ever care if the event had listeners, so just return "yes"
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}
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private async makeRecorder() {
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try {
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this.recorderStream = await navigator.mediaDevices.getUserMedia({
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audio: {
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channelCount: CHANNELS,
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noiseSuppression: true, // browsers ignore constraints they can't honour
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deviceId: MediaDeviceHandler.getAudioInput(),
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},
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});
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this.recorderContext = createAudioContext({
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// latencyHint: "interactive", // we don't want a latency hint (this causes data smoothing)
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});
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this.recorderSource = this.recorderContext.createMediaStreamSource(this.recorderStream);
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// Set up our worklet. We use this for timing information and waveform analysis: the
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// web audio API prefers this be done async to avoid holding the main thread with math.
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const mxRecorderWorkletPath = document.body.dataset.vectorRecorderWorkletScript;
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if (!mxRecorderWorkletPath) {
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// noinspection ExceptionCaughtLocallyJS
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throw new Error("Unable to create recorder: no worklet script registered");
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}
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// Connect our inputs and outputs
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if (this.recorderContext.audioWorklet) {
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await this.recorderContext.audioWorklet.addModule(mxRecorderWorkletPath);
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this.recorderWorklet = new AudioWorkletNode(this.recorderContext, WORKLET_NAME);
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this.recorderSource.connect(this.recorderWorklet);
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this.recorderWorklet.connect(this.recorderContext.destination);
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// Dev note: we can't use `addEventListener` for some reason. It just doesn't work.
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this.recorderWorklet.port.onmessage = (ev) => {
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switch (ev.data['ev']) {
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case PayloadEvent.Timekeep:
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this.processAudioUpdate(ev.data['timeSeconds']);
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break;
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case PayloadEvent.AmplitudeMark:
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// Sanity check to make sure we're adding about one sample per second
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if (ev.data['forIndex'] === this.amplitudes.length) {
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this.amplitudes.push(ev.data['amplitude']);
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this.liveWaveform.pushValue(ev.data['amplitude']);
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}
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break;
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}
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};
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} else {
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// Safari fallback: use a processor node instead, buffered to 1024 bytes of data
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// like the worklet is.
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this.recorderProcessor = this.recorderContext.createScriptProcessor(1024, CHANNELS, CHANNELS);
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this.recorderSource.connect(this.recorderProcessor);
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this.recorderProcessor.connect(this.recorderContext.destination);
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this.recorderProcessor.addEventListener("audioprocess", this.onAudioProcess);
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}
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this.recorder = new Recorder({
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encoderPath, // magic from webpack
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encoderSampleRate: SAMPLE_RATE,
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encoderApplication: 2048, // voice (default is "audio")
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streamPages: true, // this speeds up the encoding process by using CPU over time
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encoderFrameSize: 20, // ms, arbitrary frame size we send to the encoder
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numberOfChannels: CHANNELS,
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sourceNode: this.recorderSource,
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encoderBitRate: BITRATE,
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// We use low values for the following to ease CPU usage - the resulting waveform
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// is indistinguishable for a voice message. Note that the underlying library will
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// pick defaults which prefer the highest possible quality, CPU be damned.
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encoderComplexity: 3, // 0-10, 10 is slow and high quality.
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resampleQuality: 3, // 0-10, 10 is slow and high quality
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});
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this.recorder.ondataavailable = (a: ArrayBuffer) => {
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const buf = new Uint8Array(a);
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const newBuf = new Uint8Array(this.buffer.length + buf.length);
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newBuf.set(this.buffer, 0);
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newBuf.set(buf, this.buffer.length);
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this.buffer = newBuf;
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};
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} catch (e) {
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console.error("Error starting recording: ", e);
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if (e instanceof DOMException) { // Unhelpful DOMExceptions are common - parse them sanely
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console.error(`${e.name} (${e.code}): ${e.message}`);
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}
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// Clean up as best as possible
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if (this.recorderStream) this.recorderStream.getTracks().forEach(t => t.stop());
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if (this.recorderSource) this.recorderSource.disconnect();
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if (this.recorder) this.recorder.close();
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if (this.recorderContext) {
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// noinspection ES6MissingAwait - not important that we wait
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this.recorderContext.close();
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}
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throw e; // rethrow so upstream can handle it
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}
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}
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private get audioBuffer(): Uint8Array {
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// We need a clone of the buffer to avoid accidentally changing the position
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// on the real thing.
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return this.buffer.slice(0);
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}
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public get liveData(): SimpleObservable<IRecordingUpdate> {
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if (!this.recording) throw new Error("No observable when not recording");
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return this.observable;
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}
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public get isSupported(): boolean {
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return !!Recorder.isRecordingSupported();
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}
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public get hasRecording(): boolean {
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return this.buffer.length > 0;
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}
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private onAudioProcess = (ev: AudioProcessingEvent) => {
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this.processAudioUpdate(ev.playbackTime);
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// We skip the functionality of the worklet regarding waveform calculations: we
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// should get that information pretty quick during the playback info.
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};
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private processAudioUpdate = (timeSeconds: number) => {
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if (!this.recording) return;
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this.observable.update({
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waveform: this.liveWaveform.value.map(v => clamp(v, 0, 1)),
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timeSeconds: timeSeconds,
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});
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// Now that we've updated the data/waveform, let's do a time check. We don't want to
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// go horribly over the limit. We also emit a warning state if needed.
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//
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// We use the recorder's perspective of time to make sure we don't cut off the last
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// frame of audio, otherwise we end up with a 1:59 clip (119.68 seconds). This extra
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// safety can allow us to overshoot the target a bit, but at least when we say 2min
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// maximum we actually mean it.
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//
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// In testing, recorder time and worker time lag by about 400ms, which is roughly the
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// time needed to encode a sample/frame.
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//
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// Ref for recorderSeconds: https://github.com/chris-rudmin/opus-recorder#instance-fields
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const recorderSeconds = this.recorder.encodedSamplePosition / 48000;
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const secondsLeft = TARGET_MAX_LENGTH - recorderSeconds;
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if (secondsLeft < 0) { // go over to make sure we definitely capture that last frame
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// noinspection JSIgnoredPromiseFromCall - we aren't concerned with it overlapping
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this.stop();
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} else if (secondsLeft <= TARGET_WARN_TIME_LEFT) {
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Singleflight.for(this, "ending_soon").do(() => {
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this.emit(RecordingState.EndingSoon, { secondsLeft });
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return Singleflight.Void;
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});
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}
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};
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public async start(): Promise<void> {
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if (this.lastUpload || this.hasRecording) {
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throw new Error("Recording already prepared");
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}
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if (this.recording) {
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throw new Error("Recording already in progress");
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}
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if (this.observable) {
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this.observable.close();
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}
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this.observable = new SimpleObservable<IRecordingUpdate>();
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await this.makeRecorder();
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await this.recorder.start();
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this.recording = true;
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this.emit(RecordingState.Started);
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}
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public async stop(): Promise<Uint8Array> {
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return Singleflight.for(this, "stop").do(async () => {
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if (!this.recording) {
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throw new Error("No recording to stop");
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}
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// Disconnect the source early to start shutting down resources
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await this.recorder.stop(); // stop first to flush the last frame
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this.recorderSource.disconnect();
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if (this.recorderWorklet) this.recorderWorklet.disconnect();
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if (this.recorderProcessor) {
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this.recorderProcessor.disconnect();
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this.recorderProcessor.removeEventListener("audioprocess", this.onAudioProcess);
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}
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// close the context after the recorder so the recorder doesn't try to
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// connect anything to the context (this would generate a warning)
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await this.recorderContext.close();
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// Now stop all the media tracks so we can release them back to the user/OS
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this.recorderStream.getTracks().forEach(t => t.stop());
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// Finally do our post-processing and clean up
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this.recording = false;
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await this.recorder.close();
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this.emit(RecordingState.Ended);
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return this.audioBuffer;
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});
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}
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/**
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* Gets a playback instance for this voice recording. Note that the playback will not
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* have been prepared fully, meaning the `prepare()` function needs to be called on it.
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*
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* The same playback instance is returned each time.
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*
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* @returns {Playback} The playback instance.
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*/
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public getPlayback(): Playback {
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this.playback = Singleflight.for(this, "playback").do(() => {
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return new Playback(this.audioBuffer.buffer, this.amplitudes); // cast to ArrayBuffer proper;
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});
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return this.playback;
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}
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public destroy() {
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// noinspection JSIgnoredPromiseFromCall - not concerned about stop() being called async here
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this.stop();
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this.removeAllListeners();
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Singleflight.forgetAllFor(this);
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// noinspection JSIgnoredPromiseFromCall - not concerned about being called async here
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this.playback?.destroy();
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this.observable.close();
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}
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public async upload(inRoomId: string): Promise<IUpload> {
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if (!this.hasRecording) {
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throw new Error("No recording available to upload");
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}
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if (this.lastUpload) return this.lastUpload;
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try {
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this.emit(RecordingState.Uploading);
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const { url: mxc, file: encrypted } = await uploadFile(this.client, inRoomId, new Blob([this.audioBuffer], {
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type: this.contentType,
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}));
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this.lastUpload = { mxc, encrypted };
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this.emit(RecordingState.Uploaded);
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} catch (e) {
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this.emit(RecordingState.Ended);
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throw e;
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}
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return this.lastUpload;
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}
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}
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